This invention relates to a content supply system and an information processing method for supplying contents such as audio signals or image signals, and particularly to a content supply system and an information processing method in which signals are coded to enable trial viewing/listening and therefore reproduction and recording with high quality is made possible by adding a small quantity of data when a trial viewer/listener decides to purchase the signals.
A content (software) distribution method is known in which, for example, an acoustic signal or the like is encrypted and then broadcast or recorded to a recording medium so that only a person who purchased a key for decryption is permitted to listen to the signal.
As an encryption method, for example, a method is known in which an initial value of a random-number sequence is given as a key signal for a bit string of a PCM acoustic signal and then a bit string obtained by taking an exclusive OR between the generated random-number sequence of 0/1 and the PCM bit string is transmitted or recorded to a recording medium. As this method is used, a person who acquired the key signal can correctly reproduce the acoustic signal and a person who did not acquire the key signal can only reproduce noise. Of course, it is also possible to use a more complicated method such as so-called DES (Data Encryption Standard) as an encryption method. Description of the DES standard is disclosed in xe2x80x9cFederal Information Processing Standards Publication 46, Specifications for the DATA ENCRYPTION STANDARD, 1977, January 15.xe2x80x9d
On the other hand, a method for compressing an acoustic signal and then broadcasting or recording the compressed acoustic signal to a recording medium is popularized, and recording media which enable recording of a coded audio signal or the like, such as a magneto-optical disc, are broadly used.
There are various techniques for high-efficiency coding of an audio signal, voice signal, or the like. For example, such techniques may include subband coding (SBC), which is a non-blocked frequency band division system for dividing an audio signal or the like on the time axis into a plurality of frequency bands without blocking and then coding the band-divided audio signal, and so-called transform coding, which is a blocked frequency band division system for transforming (spectrally transforming) a signal on the time axis to a signal of the frequency axis, then dividing the signal into a plurality of frequency bands and coding the signal of each band. Moreover, a high-efficiency coding technique combining the above-described subband coding with transform coding is considered. In that case, for example, after frequency band division is carried out by the above-described subband coding, the signal of each band is spectrally transformed to a signal on the frequency axis and the spectrally transformed signal of each band is coded.
As a filter for the above-described technique, for example, a QMF filter is used. The QMF filter is described in xe2x80x9cR. E. Crochiere, Digital coding of speech in subbands, Bell Syst. Tech. J. Vol.55, No.8, 1976.xe2x80x9d Moreover, a filter division technique with equal bandwidth is disclosed in xe2x80x9cJoseph H. Rothweiler, Polyphase Quadrature Filtersxe2x80x94A new subband coding technique, ICASSP 83, BOSTON.xe2x80x9d
As the above-described spectral transform, for example, the time axis is transformed to the frequency axis by blocking an input audio signal by predetermined unit time (frame) and then performing discrete Fourier transform (DFT), discrete cosine transform (DCT), modified discrete cosine transform (MDCT) or the like on each of the blocks. MDCT is described in xe2x80x9cJ. P. Princen, A. B. Bradley, Univ. of Surrey Royal Melbourne Inst. of Tech., Subband/Transform Coding Using Filter Band Designs Based on Time Domain Aliasing Cancellation, ICASSP, 1987xe2x80x9d.
If the above-described DFT or DCT is used as a method for transforming a waveform signal to the spectrum, M independent real-number data are provided by performing transform on a time block consisting of M samples. To reduce the connection distortion between time blocks, each time block is usually overlapped with both adjacent blocks by M1 samples each. Therefore, on average, M real-number data are quantized and coded for (M-M1) samples in DFT or DCT.
On the other hand, if the above-described MDCT is used as a method for transforming a waveform signal to the spectrum, M independent real-number data are provided from 2M samples as a result of overlapping both adjacent time blocks by M samples each. Therefore, on average, M real-number data are quantized and coded for M samples in MDCT. A decoding device can reconstruct the waveform signal by performing inverse transform on each block of the code obtained by using MDCT and then adding the resulting waveform elements while letting them interfere with each other.
Generally, by elongating a time block for transform, the frequency resolution of the spectrum is enhanced and the energy concentrates at a specific spectral component. Therefore, by using MDCT in which each block is overlapped with both adjacent blocks by half each to perform transform with a longer block length and in which the number of resulting spectral signals is not increased from the number of the original time samples, more efficient coding can be carried out than when DFT or DCT is used. Moreover, by having each block have a sufficient long overlap with the adjacent blocks, the distortion between the blocks of the waveform signal can be reduced.
By quantizing the signal thus divided to each band by the filter or spectral transform, a band where quantization noise is generated can be controlled and more auditorily efficient coding can be performed by utilizing characteristics such as masking effect. By carrying out normalization for each band with the maximum value of absolute values of signal components in the band before performing quantization, more efficient coding can be performed.
The frequency division width in the case of quantizing each frequency component obtained by frequency band division is determined, for example, in consideration of the human auditory characteristic. Specifically, an audio signal may be divided into a plurality of bands (for example 25 bands) with broader bandwidths for higher-frequency bands which are generally called critical bands. When coding data of each band in this case, coding is carried out by using predetermined bit distribution to each band or adaptive bit allocation to each band. For example, when coding coefficient/factor data resulting from the above-described MDCT processing by the above-described bit allocation, the MDCT coefficient/factor data of each band resulting from MDCT of each of the blocks is coded by using an adaptive number of allocated bits.
For such bit allocation, the following two techniques are known. Specifically, xe2x80x9cR. Zelinski and P. Noll, Adaptive Transform Coding of Speech Signals, IEEE Transactions of Acoustics, Speech, and Signal Processing, vol.ASSP-25, No.2, August 1977xe2x80x9d, discloses bit allocation based on the magnitude of a signal of each band. In this system, though the quantization noise spectrum is flat and the noise energy is minimum, the actual perception of noise is not optimum because the auditory masking effect is not utilized. xe2x80x9cM. A. Kransner, MIT, The critical band coderxe2x80x94digital encoding of the perceptual requirements of the auditory system, ICASSP 1980xe2x80x9d, discloses a technique in which a necessary signal-to-noise ratio for each band is obtained using auditory masking, thus performing fixed bit allocation. With this technique, however, even when measuring characteristics by using a sine wave input, a satisfactory characteristic value is not obtained because of fixed bit allocation.
To solve these problems, a high-efficiency coding device is proposed in which all the bits that can be used for bit allocation are divisionally used for a fixed bit allocation pattern predetermined for each small block and for bit allocation dependent on the magnitude of the signal of each block and in which the division ratio depends on a signal related to the input signal so that the proportion of division to the fixed bit allocation pattern is increased for a smoother spectrum of the signal.
According to this technique, if the energy concentrates at a specific spectrum, as in a sine wave input, the signal-to-noise ratio can be significantly improved as a whole by allocating many bits to a block containing that spectrum. Generally, since the human auditory sense is very sensitive to a signal having an acute spectral component, the improvement in the signal-to-noise ratio by using this technique is effective not only for improvement in the numerical value in measurement but also for improvement in the sound quality perceived by the auditory sense.
Many other techniques for bit allocation are proposed. As the auditory model is elaborated further and the coding device has a sufficient capability, more auditorily efficient coding is made possible. In these techniques, typically, a bit allocation reference value of a real number which realizes the signal-to-noise characteristic found by calculation with high fidelity is found, and an integer which approximates the reference value is used as the number of allocated bits.
In the specification and drawings of the Japanese Patent Application No.H5-152865 or WO94/28633 proposed by the present inventors, a method is proposed in which a tonal component that is particularly important in terms of the auditory sense, that is, a signal component with energy concentrated around a specific frequency, is separated from a spectral signal and coded separately from the other spectral components. This enables efficient coding of an audio signal etc. at a high compression rate while causing little auditory deterioration.
To construct an actual code string, first, quantization accuracy information and normalization factor (coefficient) information may be coded using a predetermined number of bits for each band on which normalization and quantization are performed, and then a normalized and quantized spectral signal may be coded. Moreover, the ISO/IEC 11172-3:1993(E), 1993 describes a high-efficiency coding system in which the number of bits representing quantization accuracy information is set to vary depending on the band. It is standardized that the number of bits representing quantization accuracy information is decreased toward higher frequency bands.
Instead of directly coding quantization accuracy information, a method is known in which quantization accuracy information is decided from normalization factor information in a decoding device. In this method, however, the relation between normalization factor information and quantization accuracy information is decided when the standard is set. Therefore, control of the quantization accuracy based on a more advanced auditory model cannot be introduced in the future. Moreover, if the compression rate to be realized is variable, the relation between normalization factor information and quantization accuracy information must be defined for each compression rate.
There is also known a method for more efficient coding by coding a quantized spectral signal using a variable-length code described in xe2x80x9cD. A. Huffman, A Method for Construction of Minimum Redundancy Codes, Proc. I.R.E., 40, p.1098 (1952)xe2x80x9d.
Meanwhile, a software distribution method is known in which an acoustic signal or the like coded by the above-described methods is encrypted and broadcast or recorded to a recording medium so that only a person who purchased a key is permitted to listen to the signal. As an encryption method, for example, a method is known in which an initial value of a random-number sequence is given as a key signal for a bit string of a PCM (pulse code modulation) acoustic signal or a coded signal and then a bit string obtained by taking an exclusive OR between the generated random-number sequence of 0/1 and the bit string is transmitted or recorded to a recording medium. As this method is used, only a person who acquired the key signal can correctly reproduce the acoustic signal and a person who did not acquire the key signal can only reproduce noise. Of course, it is also possible to use a more complicated method as an encryption method.
However, such a method has a problem that a user must first decide to purchase a content and then download a large quantity of data with high quality from a content supply center or the like.
There is another problem that a user must necessarily access the content supply center or the like in order to carry out trial viewing/listening and hence there will be less opportunities of trial viewing/listening.
Moreover, in the case of music or the like, a user often wants to purchase a tune selected from tunes which the user happens to listen to. However, in the method of accessing the center to select trial viewing/listening, there is a problem that the user will be at a loss about which tune to select for trial listening.
In view of the foregoing status of the art, it is an object of the present invention to provide a content supply system and an information processing method which enable trial viewing/listening, eliminate the need to download a large quantity of data with high quality from a content supply center or the like, and enable increase in the number of opportunities of content purchase.
In order to solve the foregoing problems, a content supply system according to the present invention comprises: a content supply center for supplying a signal of a first code string in which a part of information of a code string obtained by coding a content is replaced by dummy data and a signal of a second code string for complementing the dummy data part; and a user terminal having a function to arbitrarily receive the data of the first code string from the content supply center, to receive the signal of the second code string from the content supply center in accordance with a predetermined condition, and to replace the dummy data of the first code string by using the second code string.
Thus, it is no longer necessary to download a large quantity of data with high quality after deciding to purchase the content.
Moreover, a content supply system according to the present invention comprises a content supply center for sending signals of contents for trial viewing/listening to a user terminal for free or at a low price, a user being able to receive the contents for trial viewing/listening from the content supply center and to select and purchase only the content that the user likes from the received contents so as to reproduce the content with high quality.
Thus, since the data for trial viewing/listening is automatically sent from the center, the user need not access the center for trial viewing/listening and there will be more opportunities of trial viewing/listening.
Moreover, a content supply system according to the present invention permits copying of signals of contents for trial viewing/listening acquired by one user, for free or at a low price, and enables each user who reproduced the copy to select and purchase only the content that each user likes from the reproduced copy so as to reproduce the content with high quality.
Thus, since the trial viewing/listening data is transmitted between the users, there will be more opportunities of trial viewing/listening.
Furthermore, a content supply system according to the present invention enables a user to record signals of contents for trial viewing/listening downloaded from a content supply terminal for free or at a low price, reproduce the signals of these contents, and select and purchase only the content that the user likes from them so as to reproduce the content with high quality.
Thus, since the trial viewing/listening data is transmitted between the users, there will be more opportunities of trial viewing/listening.
The signals of the contents for trial viewing/listening have a narrow reproducing band. The narrowing of the reproducing band is realized by embedding dummy data into the code string of the content or by encrypting a part of the code string.
Moreover, according to the present invention, since a user more often purchases a music tune selected from tunes which the user happens to listen to, the user will not be at a loss about which tune to select for trial listening and there will be more opportunities of content purchase.
Furthermore, an information processing method according to the present invention comprises: a step of generating, from an original code string obtained by coding original information by a predetermined format with a frame structure, a first code string which can be reproduced as information having lower quality than the original information but recognizable by a human being, by replacing a part or a plurality of parts of the original code string with dummy data; a step of generating a second code string which enables reproduction of the original information by complementing the first code string, from the part or the plurality of parts of the original code string separated from the first code string; a step of supplying the first code string to first information supply means; a step of supplying the second code string to the first information supply means or second information supply means; a step of distributing the first code string through the first information supply means so that at least a part of the first code string is in an access-free status; a step of distributing the second code string through the first information supply means or the second information supply means so that at least a part of the second code string is in a non-access-free status; a step of reproducing the information having lower quality than the original information but recognizable by a human being from the first code string, by using information reproducing means capable of reproducing the code string coded by the predetermined format; and a step of accessing the second code string through predetermined processing and reproducing the information from the first code string while complementing the first code string with the second code string.
The part or the plurality of parts of the code string to be replaced by the dummy data may include, for example, a part or all of content information and/or a part or all of control information necessary for reproduction of the content information. Moreover, the part or the plurality of parts of the code string to be replaced can be set to a part or all of frames. Furthermore, the predetermined format may be a format for generating a code string, for example, by variable-length coding the content information and multiplexing it as a code string related to control information necessary for reproduction of the content information.
The access-free status means that an unspecified person can normally access the code string. That is, the access-free status includes not only the status where no access limitation is set on the code string but also the status where the code string is practically access-free though an access limitation that can be easily canceled by an unspecified person is set. The non-access-free status means that only a specified person can normally access the code string.
The distribution in the non-access-free status can be realized, for example, by performing encryption processing on the code string and then distributing the code string. In this case, the user cannot access the code string unless he/she carries out decryption processing as predetermined processing.
Moreover, the distribution in the non-access-free status can be realized, for example, by recording the code string to a copy-protected storage medium and physically distributing the storage medium. In this case, free access to the code string by the user for a predetermined period may be permitted by utilizing a digital watermarking technique, a time limitation technique and the like.
In any of the above-described methods for realizing the distribution in the non-access-free status, it is possible to employ such a structure that execution of the corresponding predetermined processing by the user is permitted on the assumption of appropriate accounting, registration of user information, distribution of advertising information to the user or confirmation of advertising information by the user, and so on. Alternatively, in any of the above-described methods for realizing the distribution in the non-access-free status, it is possible to employ such a structure that free execution of the predetermined processing by the user within a predetermined period is permitted but free execution of the predetermined processing by the user is not permitted after the predetermined period unless appropriate processing is carried out.